| Add_workflow_for_techpubs |
|---|
| AUTH1 | UserResourceIdentifier{userKey=8a00a0c85b2726c2015b58aa779d0003, userName='null'} |
|---|
| JIRAIDAUTH | CHOR-1718 |
|---|
| REV5 | UserResourceIdentifier{userKey=8a00a0c85b2726c2015b58aa779d0003, userName='null'} |
|---|
| REV6 | UserResourceIdentifier{userKey=8a00a0c85b2726c2015b58aa779d0003, userName='null'} |
|---|
| REV3 | UserResourceIdentifier{userKey=8a00a02355cd1c2f0155cd26cef30cd0, userName='null'} |
|---|
| REV1 | UserResourceIdentifier{userKey=8a00a0c857f0393d01581c58d2400005, userName='null'} |
|---|
|
| CSS Stylesheet |
|---|
img.confluence-embedded-image { display: inline-block !important; } |
To create or modify a Voice Codec Profile:
...
From the Create Media Profile drop down box at the top of the Media Profiles page, select Voice Codec Profile.Complete Voice Codec Configuration fields (see Field Definitions below), and click OK.
From the Codec drop down box, select a codec.
Configure the files and click OK. See field definitions below.
| Panel |
|---|
|
| Caption |
|---|
| 0 | Figure |
|---|
| 1 | Create Voice Codec Profile |
|---|
|  Image Modified
|
|
Voice Codec Profiles - Field Definitions
| Caption |
|---|
|
| Parameter | Description |
|---|
| Description | A name/description for the codec. | | Codec | Specifies the voice codec and encoding scheme used towards the IP side of a VoIP call. The |
|
chosen selected codec affects the audio quality and bandwidth consumption of VoIP calls to which you apply |
|
this the Voice Codec Profile (in the Media Profiles List |
|
Profile for codec will depend depends on the interoperability requirements for connecting to other voice peers |
|
, as well as on the and bandwidth requirements. Most codecs use data compression algorithms, which saves |
|
the , it slightly reduces the voice quality |
|
, whereas . G.711 does not use compression and |
|
therefore most bandwidth.Note: The G.722.2 Codec only appears in the list box if the AMR license is installed.
...
more bandwidth. The following codecs are supported |
|
on SBC 1000/2000 platform:...
...
platform:- G.711 A-Law
- G.711 u-Law
- G.722
- G.722.2
- G.723.1
- G.726
|
|
...
...
Codec - Additional Specifications - G.722 Wideband Codec: Supports 20ms packet sized and a bit-rate of 64Kbps only.
- G.722 or AMR-WB channels: Does not support fax data detection support.
- G.722.2: Appears only if the AMR License is installed.
- G.726: Supports
|
|
...
...
- G.729: Supports G.729A and G.729AB
|
|
...
- . The variant used when selecting
|
|
...
...
- depends upon the setting of Media List flag
|
|
...
...
- . If Silence Suppression is enabled, G.729AB is used.
- Opus: Supported in SBC SWe Lite only.
NOTE: Applicable to SWe Lite initiating FAX setup. A G.711 codec must be used for fax detection; tones are detected on the G711 egress (answering CED tone) leg. Otherwise the peer has to initiate fax path. | | Rate | Specifies the voice sampling rate in bits/sec |
|
to be setting parameter applies to G.722.1, G.723.1 |
|
and , G.726, and Opus codecs only. |
|
For example: the codec sample interval for G.711A-Law and G.711u-Law is 10 ms.Therefore, the voice payload sizes are For all other codecs, the voice sampling rate is fixed and defined in the appropriate specification for that codec. | | Payload Size |
|
Specifies (indirectly) the number of bytes in a single packet. The payload size is a multiple of the codec sample size.
The recommended length of time in milliseconds rounded up to the next full integer value represented by the media in a packet. Valid entry: 10, 20, 30, 40 |
|
, 50. The codec sample interval for G.723.1 is 30 ms and therefore the payload sizes are 30, 60, 90, 80, 90 (specific codec determines values available). - Smaller payload sizes decrease audio transport latency at the expense of higher bandwidth consumption.
- Larger packet sizes reduce the
|
|
use of - bandwidth. The larger the payload size, the fewer and larger the packets.
|
|
When - With larger payload sizes
|
|
are used headers as well as fewer - and IP/UDP/RTP headers are used
|
|
. The - ; a disadvantage is that if UDP packets get lost, the impact on the voice quality
|
|
will be much - is higher because a single packet contains more raw voice data
|
|
.Total packet size = (L2 header: MP or FRF12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)...
For Re-Invite Only: If the SBC receives a larger than configured payload size from the peer offer in the re-invite, the SBC rejects with a 488 'Not Acceptable Here' response. The call rolls back to the previous negotiated offer answer. | | Payload Type | Specifies the payload type for this profile. |
|
Only available when either and is the specified codec, and Opus codecs only. | Payload Format
| Specifies the |
|
payload format for this profile. This field is only available when the mode for the payload: Bandwidth Efficient Mode or Octet Aligned Mode. Default entry: Bandwidth Efficient Mode. This setting applies to G.722.2 |
|
codec is selected- Bandwidth-Efficient Mode
- Octet-Aligned Mode
For a given session, the payload format can be either bandwidth efficient or octet aligned, depending on the mode of operation that is established for the session via out-of-band means. In the octet-aligned format, all the fields in a payload, including payload header, table of contents entries, and speech frames themselves, are individually aligned to octet boundaries to make implementations efficient. In the bandwidth-efficient format, only the full payload is octet aligned, so fewer padding bits are added.
| Voice Bit Rate | The maximum Opus bit rate (in bits/second) used for the current session. Valid entry: 0 (use variable bit rate) or 1 (use constant bit rate). Default entry: CBR. Parameter applies to Opus only. | | Use FEC | Specifies whether to use Forward Error Correction (FEC). Valid entry: 0 (do not use FEC) or 1 (use FEC). Default entry: 0. Parameter applies to Opus only.
| | Use DTX | Specifies whether to use Discontinuous Transmission (DTX). Valid entry: 0 (do not use DTX) or 1 (use DTX). Default entry: 0. Parameter applies to Opus only.
|
|
...